- sr-users - kamailio.org

IMS - change the transport layer protocol for Diameter based interfaces (especially Rx)
by "björn bauer" 05 Jun '19

05 Jun '19

3 2

RTPEngine and dialog moving.
by Igor Olhovskiy 04 Jun '19

04 Jun '19

Hi! Wanted to clarify some info about rtpengine working within HA-environment. Idea is we have multiple rtpengines under same setid. So, when I'm calling rtpengine_offer() I really don't care which is selected. But than might happen situation, where this call(dialog) is moved to other Kamailio instance (via dlg_db_load_callid()) and from there called rtpengine_delete(). So, my question is, how I can be sure, that delete request will go to same rtpengine instance that was selected in rtpengine_offer() on different server. And where this info is stored, within dialog or within TM module (which is not replicated in any way) Regards, Igor

3 4

Info: app_lua_sr module - old Lua sr library
by Daniel-Constantin Mierla 04 Jun '19

04 Jun '19

Hello, several days ago I split the app_lua module, relocating the code related to old Lua sr library to app_lua_sr module, keeping in this way app_lua smaller and focused to the KEMI code. Over the time, app_lua_sr should be deprecated and removed, as we are sure the functions in Lua sr have an equivalent in Lua KSR. Therefore, if you are using in Lua scripts functions from 'sr' library, consider migrating to KSR as soon as possible. The intermediary solution for a quick upgrade to master version (to be release as v5.3.x) is to load app_lua_sr and update some modparams, like: loadmodule "app_lua.so" loadmodule "app_lua_sr.so" modparam("app_lua_sr", "register", "sl") If you find a function from Lua sr not available in Lua KSR, then report it via mailing lists or bug tracker. An article about this topic was published at: * https://www.kamailio.org/w/2019/05/new-module-app_lua_sr/ Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Multiple definitions with OR or AND in "ifdef" in kamailio script
by Maxim M 04 Jun '19

04 Jun '19

Is it possible to do something like this in kamailio script?: !!ifdef DEF1 || DEF2 .... !!endif

2 1

empty action lists in route block
by Maryam Baghdadi 04 Jun '19

04 Jun '19

Hello everyone! I tried to use from Lua scripting language in kamailio configuration and rewrite all routes in Lua script. and I export many functions in ims_auth in KEMI and recompile ims_auth module. I have a problem with www_ims_challenge function calling in Lua script. when I define and call a function (REG_MAR_ROUTE) in www_ims_challenge as a route black, I get this error " empty action lists in route block" . why this happens and what's the solution ? thanks in advance

3 2

Removal of Dispatcher
by Colin, Andrew 04 Jun '19

04 Jun '19

I am using the API to remove dispatchers with a simple curl command, is there a way i can make kamailio check that there are sufficient dispatchers available before removing them when the command is run? Example: i am removing asterisk1 out of kamailio dispatcher list, but kamailio needs to ensure that there is atleast 2 or 3 other asterisk servers active in the list before it allows it to be removed. any ideas how i can achieve this?

2 3

AAR change subscription ID
by eyas barhouk 03 Jun '19

03 Jun '19

dear community, when i'm trying to deploy kamailio IMS with PCRF i get ims-qos AAR error due to Subscription id. in ims_qos module i found that i can change the subscription_id type and value as following: * subscription_id to hard code subscription ID for AAR. Used for some broken PCRFs. Leave blank to use default * subscription_id_type to hard code subscription ID type for AAR. Only applicable if subscription_id is set. Set to -1 to use default. This is as per RFC 4006: END_USER_E164 0, END_USER_IMSI 1, END_USER_SIP_URI 2, END_USER_NAI 3, END_USER_PRIVATE 4 but when i changed it there was no change on AAR Subscription-id AVP. any advise about how to set or change the subscription id dears ?? B.R Eyas

1 0

Releasing a new version for Siremis
by Daniel-Constantin Mierla 03 Jun '19

03 Jun '19

Hello, after quite a bit of time, we are considering to release a new version for Siremis - the web management interface for Kamailio. A lot of effort has been pushed to address the issues with PHPv7.x, which deprecated a lot of PHPv5.x API as well as made more strict rules in regards to classes and inheritances. At this stage it should be in good stage, but it would help if others can give it a ride to Siremis master branch: * https://github.com/asipto/siremis Installation guidelines: * https://kb.asipto.com/siremis:install51x:main Testing with both PHP v5 and v7 is appreciated in order to be sure things were not broken for old systems. Any issue found should be reported on github project issue tracker. If all goes fine, in soon we will make the release. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

invalid payload location
by Kelvin Chua 31 May '19

31 May '19

is there a plan to create an sdp sanity functionality? ERROR: <core> [core/parser/sdp/sdp.c:212]: set_sdp_payload_attr(): Invalid payload location opensips has something similar sipmsg_validate() *'s'* - checks the integrity of the SDP body, if it exists Kelvin Chua

2 2

Table creation error when enabling ims_usrloc_scscf module.
by Ali Shirvani 30 May '19

30 May '19

Hi all, I installed kamailio v5.2.3 from source and IMS modules included. Then I tried to create required database and tables, I removed "dialog" from STANDARD_MODULES and added "ims_dialog ims_charging ims_usrloc_scscf ims_usrloc_pcscf presence" to the EXTRA_MODULES in kamctlrc file. When I issue `<kamailio_home>/sbin/kamctl create` I got following error: ERROR 1062 (23000) at line 39: Duplicate entry 'subscriber' for key 'table_name_idx' ERROR: Creating extra tables failed at ims_usrloc_scscf! Should I exclude an other module from STANDARD_MODUELS? Regards, Ali

2 1

turn on DEBUG level debugging for ICSCF module
by Trinh, Jenny 30 May '19

30 May '19

Hi, How do I turn on DEBUG level debugging for ICSCF module? Thanks Jenny

2 1

app_java.so: undefined symbol: JNI_CreateJavaVM please help
by Renato Gallo 30 May '19

30 May '19

gcj is not a requirement since 2018 [ https://www.codesd.com/item/undefined-symbol-jni-createjavavm-on-linux.html | https://www.codesd.com/item/undefined-symbol-jni-createjavavm-on-linux.html ] all details are in the bug ticket at the link: [ https://github.com/kamailio/kamailio/issues/1969 | https://github.com/kamailio/kamailio/issues/1969 ] please help Renato Gallo System Engineer sede legale e operativa: Via Privata Cefalonia, 14 - 20156 - Milano (MI) Tel. +39 02 - 87049490 Fax +39 02 - 48677349 Mobile. +39 342 - 6350524 Wi | FreeNumbers: https://freenumbers.way-interactive.com Wi | SMS: https://sms.way-interactive.com Wi | Voip: https://voip.way-interactive.com Asterweb: http://www.asterweb.org Le informazioni contenute in questo messaggio e negli eventuali allegati sono riservate e per uso esclusivo del destinatario. Persone diverse dallo stesso non possono copiare o distribuire il messaggio a terzi. Chiunque riceva questo messaggio per errore è pregato di distruggerlo e di informare immediatamente [ mailto:info@sigmaware.it | info@ ] asterweb.org

1 0

Fwd: Stack smash error when using auth_ephemeral with KEMI
by Grant Bagdasarian 30 May '19

30 May '19

Hi Daniel, See pastebin url below for the backtrace. https://pastebin.com/tTUUk4YH Hopefully I did it the right way. Please let me know if you need anything else. FYI, Kamailio is running inside of a docker container. version: kamailio 5.2.1 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144 MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 6.3.0 I can easily replicate this, but to be sure I've saved the core file locally on my machine. Regards, Grant On Thu, May 2, 2019 at 10:05 AM Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: > > Hello, > > can you get a core file for this case? > > Set `ulimit -c unlimited` before starting Kamailio as root and run again. > > Once you have the core file, get the backtrace with gdb. > > Cheers, > Daniel > > On 01.05.19 20:26, Grant Bagdasarian wrote: > > Hello, > > > > I was trying out the auth_ephemeral module inside the xhttp > > event_route in Python KEMI, and discovered a possible bug. > > > > https://pastebin.com/4Ghumfis > > > > The following will result in the stack smash error. Whenever I leave > > out the unixtimestamp the > > module will complain about not being able to convert to an int, which > > make sense, but doesn't lead to a stack smash error and a complete > > crash of Kamailio. > > > > KSR.auth_ephemeral.autheph_authenticate("1576734089:grant", "1234"). > > > > According to the documentation this module can only be used in > > REQUEST_ROUTE, but I found a presentation online which also used this > > module and that particular function in event_route, which is what I > > need for WebSocket authentication. > > > > Hope this helps. Please let me know if you need more information. > > > > Regards, > > > > Grant > > > > _______________________________________________ > > Kamailio (SER) - Users Mailing List > > sr-users(a)lists.kamailio.org > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla -- www.asipto.com > www.twitter.com/miconda -- www.linkedin.com/in/miconda > Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com >

2 10

Kamailio Load Balancing
by Gaurav Bmotra 30 May '19

30 May '19

HI i am using kamailio 5.1.2 With Ubuntu 18.04.2 LTS and using Freeswitch as Media server and kamailio as Register server i wan to implement load balance on this plz help thank you Gaurav -- *Regards:* Gaurav Kumar

3 7

usrloc extra attributes and db_mode=0
by Patrick Wakano 29 May '19

29 May '19

Hello list, Hope you all doing well! I am trying to use the extra attributes (xavp_contact) of the usrloc module to save some additional info about the user. I am setting the value before the save() and doing a call to registered() (not lookup()) before trying to access these extra attributes. This works fine with db_mode=3 but does not work in case of db_mode=0. Does anyone knows if it should also work with mode 0? I was expecting it to work but a call to kamcmd ul.dump show nothing related..... I don't want to use the DB for the location purposes because I prefer to have multiple servers using the dmq_usrloc (that works like a charm!) and it conflicts with DB persistence of the location table (when the DMQ replication happens, all servers (sharing the same DB) try to save the same user info in the location table) Thank you, Kind regards, Patrick Wakano

2 6

Statistics help
by Duarte Rocha 29 May '19

29 May '19

Greetings, I'm doing some monitoring of my Kamailio in order to quickly identify system failures and overloads. In order to do that i have scripts that read the output of "kamcmd sl.stats" and identify the rising of "500_replies", "5xx_replies" and "6xx_replies". However, i have rate limiting on my Kamailio, which generates "503" replies when limit is reached, which generates an unnecessary alarm. Is there a way to get more specific stats about the repleis generated by Kamailio? If that isn't possible, will i be safe only monitoring "500" and "6xx" replies? In my experience, when the server has reached a processing or memory limit, the reply is always 500, but i'm not really sure about this. Thank you for your help. Best Regards, Duarte Rocha

2 1

Push proxy and load balancer for asterisk?
by Gerrit Jacobsen 29 May '19

29 May '19

Good morning, I want to realise a push notification proxy to signal incoming calls from asterisk to softphones. Is Kamailio the preferred solution for this? I looked also at Flexisip and Asterisk scripts and was told by a seasoned professional that Kamailio is the preferable solution as its scalable. True? I would be glad if someone could share his experiences. Also can one Kamailio instance be used as an asterisk load balancer and also push notification proxy ? I have been a heavy openser user a long time ago. I used first openser as my core routing engine but then switched to yate. Please excuse my rather broad questions. From my experience with Openser the devil is in the detail. So any hints on where this setup may cause problems would be great. Thanks a lot for your input. Gerry

3 2

Kamailio Lord Balance
by Gaurav Bmotra 29 May '19

29 May '19

hi i m using kamailio 5.1 with Ubuntu 18.04.2 LTS and freeswitch as Media server -- *Regards:* Gaurav Kumar

4 4

RFC: split out old sr api from app_lua
by Daniel-Constantin Mierla 29 May '19

29 May '19

Hello, I am considering to split the app_lua module in order to move the older style API available under Lua 'sr' module in a dedicated module (like app_lua_sr). Later this module can be obsoleted and removed, once we know that everything there is covered by KEMI KSR Lua module. In this way, we make it more clear that KSR is the preferred API to use as well have the app_lua smaller, for those using only KSR and not needing 'sr' API. Therefore I am starting the discussion to see if anyone has something against or other suggestions. I cc-ed sr-dev to make developers aware, but I think the best place for discussion is sr-users, being related to how Kamailio is used, therefore reply only to sr-users. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

2 2

RTPEngine / Transcoding question
by Carsten Bock 29 May '19

29 May '19

Hi, I want to implement selective transcoding, e.g. avoiding Transcoding between a HD codec (G722) and a non HD Codec (G711). In case the caller does offer HD codecs, this is fine, e.g. if(sdp_with_codecs_by_name("G722,OPUS")) { rtpengine_offer("codec-transcode=OPUS codec-transcode=G722 codec-transcode=PCMA"); } else { rtpengine_manage("codec-transcode=PCMA"); } Now I have the issue, if the callee only sends me G711, I don't want to offer G722 or Opus to the caller. However, rtpengine_answer() does not seem to accept "codec-strip=G722 codec-strip=OPUS", e.g.: onreply_route() { if(!sdp_with_codecs_by_name("G722,OPUS")) { rtpengine_ manage("codec-strip=OPUS codec-strip=G722"); } } As a result, the SDP always contains the transcoding options, e.g. G722 and OPUS. I always end up in transcoding G722 to G711, which is meaningless in that case. Any ideas on how to solve that? (above examples very simplified) Thanks, Carsten -- Carsten Bock I Managing Director ng-voice GmbH Millerntorplatz 1 I 20359 Hamburg I Germany www.ng-voice.com Mobile +49 (0)179-20 21 244 I Direct +49 (0)40-52 47 593-40 I Fax +49 (0)40-52 47 593-99 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock, Dr. David Bachmann Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/

2 5

RTPEngine: rtpengine_manage(), Subsequent requests, Transcoding
by Carsten Bock 29 May '19

29 May '19

Hi, quick question: If understood it correctly in the past (and it worked for me quite well this way), I can simply call rtpengine_manage() for subsequent requests without adding any additional information (e.g. used interfaces, RTP/SRTP translation or anything like that), as this information is stored in the internal hash table of the RTPEngine module during call setup and initial offer/answer. However, this does not seem to apply to the transcoding options. When I just call "rtpengine_manage()" for a Re-INVITE of a transcoded call (in my case AMR-WB to G722), the transcoding options are no longer applied. The forwarded offer does not get the transcoded codecs added (e.g. in my case G722 is not added to the list of AMR/AMR-WB from the phone). Did I misunderstand something or is it a bug? My RTPEngine is still on 7.2, but I will double check with more recent versions today or tomorrow. However, I also found no commit messages relating to this. Thanks, Carsten -- Carsten Bock I Managing Director ng-voice GmbH Millerntorplatz 1 I 20359 Hamburg I Germany www.ng-voice.com Mobile +49 (0)179-20 21 244 I Direct +49 (0)40-52 47 593-40 I Fax +49 (0)40-52 47 593-99 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock, Dr. David Bachmann Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/

3 2

Call Routing at IMS
by Iman Mohammadi 29 May '19

29 May '19

Hi Everyone When P-CSCF receives INVITE from client it sends INVITE to S-CSCF then S-CSCF without any look up(checking the user is registered or not)sends INVITE to I-CSCF then I-CSCF sends LIR and receives LIA and sends INVITE to same S-CSCF (in case of one S-CSCF exists).My Question is here why without any look up S-CSCF sends INVITE to I-CSCF??this procedure affects system performance and increases call set up time

2 1

WebRTC bad message (offset: 22)
by Ilie Soltanici 28 May '19

28 May '19

Hello, Can someone help me to discard those messages from kamailio log files? ERROR: <core> [core/parser/parse_fline.c:262]: parse_first_line(): parse_first_line: bad message (offset: 22) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<HTTP/1.1 101 Switching Protocols#015#012Sia: SIP/2.0/TLS 212.2.161.34:60477#015#012Sec-WebSocket-Protocol: sip#015#012Upgrade: websocket#015#012Connection: upgrade#015#012Sec-WebSocket-Accept: VeiXbJClWneIrIyYgM5v3XKYHLA=#015#012Server: WebRTC GW#015#012Content-Length: 0#015#012#015#012> As I understand this is just an HTTP response to the first request, and shouldn't go through config file execution, but I don't understand why it is getting through it. Thank you

1 0

kamailio webrtc 183 early media.
by Zhan Bazarov 28 May '19

28 May '19

Hello! I have setup kamailio+rtpengine+asterisk.. And I have issue with early media for WEBrtc endpoints... As I understood right, for this case I need to use rel100(PRACK) for sending early media... But I didn't find any example for solve this issue... Can you help me with it? It would be greatly appreciated!

1 0

mem leak on 5.1.2?
by Andrew Chen 27 May '19

27 May '19

Hey guys, Need some help with possible run away memory issue on my 5.1.2 Kamailio. I noticed I'm getting this memory error from kam logs: May 24 13:47:52 ashintgtpsg51 /usr/sbin/kamailio[1857]: WARNING: http_client [functions.c:298]: curL_query_url(): Curl library out of memory (url:https://<some url>) my -m on kamailio startup is currently set to 64 but there is only 1 call going through. In our check_mk, I noticed the memory usage is ramping up: [image: image.png] What's the cause of this? -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

4 8

Getting the following error when sending INVITE request for calling
by Mohit Chawla 27 May '19

27 May '19

Hi , I am getting the following error in trying to send INVITE message from SIP client. {2 2 INVITE 4_Do2VIMWLR6Wf5ubiBxgg..} NOTICE: acc [acc.c:281]: acc_log_request(): ACC: transaction answered: timestamp=1558902843;method=INVITE;from_tag=69588468;to_tag=ebd29a60;call_id=4_Do2VIMWLR6Wf5ubiBxgg..;code=200;reason=OK;src_user=1016;src_domain=172.25.22.23;src_ip=172.25.22.40;dst_ouser=1015;dst_user=1015;dst_domain=172.25.22.40 May 27 02:04:03 epc-srv-03 /usr/sbin/kamailio[17238]: {2 2 INVITE 4_Do2VIMWLR6Wf5ubiBxgg..} ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Unknown column 'src_user' in 'field list' (1054) May 27 02:04:03 epc-srv-03 /usr/sbin/kamailio[17238]: {2 2 INVITE 4_Do2VIMWLR6Wf5ubiBxgg..} ERROR: <core> [db_query.c:243]: db_do_insert_cmd(): error while submitting query May 27 02:04:03 epc-srv-03 /usr/sbin/kamailio[17238]: {2 2 INVITE 4_Do2VIMWLR6Wf5ubiBxgg..} ERROR: acc [acc.c:477]: acc_db_request(): failed to insert into database May 27 02:04:10 epc-srv-03 /usr/sbin/kamailio[17229]: {2 3 BYE 4_Do2VIMWLR6Wf5ubiBxgg..} NOTICE: acc [acc.c:281]: acc_log_request(): ACC: transaction answered: *Can anyone help me understand the reason for this error.* *I am attaching the used kamailio.cfg file in this message.* *Thanks and Regards,* *Mohit* -- _Disclaimer: This email and any files transmitted along with it may contain Azcom confidential and proprietary information. If you are not the intended recipient, you are notified that disclosing, copying, distributing or taking any action based on the contents of the information contained herein is strictly prohibited. If you are not an intended recipient of this transmission and you received it in error, please inform the sender by reply e-mail and destroy this and all other copies of this transmission to which you have access. The sender of this email or Azcom does not accept liability for any errors or omissions in the contents of this message that may occur as a result._

2 6

remove_hf() leaves junk in the headers?
by Oded Arbel 27 May '19

27 May '19

I'm trying to rewrite the `From` header in an invite going out to an external system (*), by calling remove_hf("From"); followed by append_hf("From: <some value>\r\n"); but the result is an invalid SIP message, because apparently after `remove_hf()` finishes, it leaves the content of $fU in place of the `From` header - just the raw text, without a header name or even a "\r\n" - and that eats the next header line and breaks the INVITE. Here is how the INVITE originally looks (without either `remove_hf()` or `append_hf()`): ---8<--- INVITE sip:972522353283@10.20.30.40:5060 SIP/2.0 Record-Route: <sip:10.90.80.70;lr;did=649.f0b> Via: SIP/2.0/UDP 10.90.80.70:5060;branch=z9hG4bK6188.ff5c4fd855bd2970fdf32d4807b114c2.0 Via: SIP/2.0/UDP 10.90.80.71:5066;received=10.90.80.71;branch=z9hG4bK581897a4;rport=5066 Max-Forwards: 69 From: <sip:972515340014@example.com:5066>;tag=as0bcee217 To: <sip:972522353283@10.20.30.40:5060> Contact: <sip:972515340014@10.90.80.71:5066> Call-ID: 722299dc15aad53d28172d433e8665f4@10.90.80.71:5066 CSeq: 102 INVITE User-Agent: Asterisk PBX 14.7.2 Date: Sun, 26 May 2019 14:42:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces X-CID: ad199638-4df3-4aef-81f1-1b50581032a7 Content-Type: application/sdp Content-Length: 280 v=0 o=root 817636675 817636675 IN IP4 10.50.60.70 s=Asterisk PBX 14.7.2 c=IN IP4 10.50.60.70 t=0 0 m=audio 39064 RTP/AVP 8 0 18 9 a=maxptime:60 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:9 G722/8000 a=sendrecv a=rtcp:39065 a=ptime:20 ---8<--- and here is another call with the same setup, with just `remove_hf("From");` being invoked (without calling `append_hf()` - this is just for reproduction): ---8<--- INVITE sip:972522353283@10.20.30.40:5060 SIP/2.0 Record-Route: <sip:10.90.80.70;lr;did=f45.91b1> Via: SIP/2.0/UDP 10.90.80.70:5060;branch=z9hG4bK024d.2690ebf7a809edca2208e6b2561ccf9e.0 Via: SIP/2.0/UDP 10.90.80.71:5066;received=10.90.80.71;branch=z9hG4bK79ae3073;rport=5066 Max-Forwards: 69 972515340014To: <sip:972522353283@10.20.30.40:5060> Contact: <sip:972515340014@10.90.80.71:5066> Call-ID: 7b20ba5a549f6f886c22985146765736@10.90.80.71:5066 CSeq: 102 INVITE User-Agent: Asterisk PBX 14.7.2 Date: Sun, 26 May 2019 14:44:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces X-CID: ffbb4ac1-3a0f-4dd6-9ce9-7ad24361081a Content-Type: application/sdp Content-Length: 280 v=0 o=root 509740923 509740923 IN IP4 10.50.60.70 s=Asterisk PBX 14.7.2 c=IN IP4 10.50.60.70 t=0 0 m=audio 39102 RTP/AVP 8 0 18 9 a=maxptime:60 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:9 G722/8000 a=sendrecv a=rtcp:39103 a=ptime:20 ---8<--- As you can see, the `From` header is indeed gone, but it left the $fU just before the `To` header. WTF? I'm running Kamailio 5.2.2 built locally. Notes: *) the problem I'm trying to solve is that I can't get my Asterisk installation to generate the `From` header I want, the reason is out of scope of this question, so I'm sending custom SIP headers over to Kamaiio and I want to rewrite the header in Kamailio to the "correct" value. -- Oded Arbel

6 8

PRE-PAY update timer
by Steve Bucklin 25 May '19

25 May '19

Hello all (again), I am looking at a 'good' method to start a call with "chunks" of time allocated. I wish the initial call to check a credit, and if credit is good, allow a small amount of time (say 30 seconds). I have enabled DIALOG and have looked at CNXCC, but am confused! When the initial 30 seconds runs out, I want to authorise another 30 seconds, etc, etc, etc. I have started to play and set a dialog timer that expires to a route, and then can 'return' allowing the call to continue - BUT was not able to reset the dialog timer to trip again after another 30 seconds. I tried to 'set_max_time' on cnxcc, but this did not seem to trip the event route upon timeout? Does anyone have a pointer to any examples? I also am unsure of how to update the timer and keep a call in play! Steve

5 4

No Private memory left using usrloc with db_redis in mode 3
by José Seabra 24 May '19

24 May '19

Hello there, i have stored ~200 000 registrations contacts in redis database and when kamailio starts looking for expired contacts in redis then it starts giving these errors: WARNING: PR_LOG: 123: db_redis [redis_dbase.c:1099]: db_redis_perform_query(): performing full table scan on table 'location' while performing query WARNING: PR_LOG: 123: db_redis [redis_dbase.c:1102]: db_redis_perform_query(): scan key 0 is 'expires' *WARNING: PR_LOG: 123: db_redis [redis_dbase.c:1102]: db_redis_perform_query(): scan key 1 is 'expires'* *ERROR: PR_LOG: 123: <core> [core/mem/q_malloc.c:291]: qm_find_free(): qm_find_free(0x7ff80922b010, 3055920); Free fragment not found!* *ERROR: PR_LOG: 123: <core> [core/mem/q_malloc.c:425]: qm_malloc(): qm_malloc(0x7ff80922b010, 3055920) called from core: db_res.c: db_allocate_rows(183), module: core; Free fragment not found!* *ERROR: PR_LOG: 123: <core> [db_res.c:185]: db_allocate_rows(): no private memory left* *ERROR: PR_LOG: 123: db_redis [redis_dbase.c:1115]: db_redis_perform_query(): Failed to allocate memory for rows * I have configured the private memory to 128 MB but seems that isn't enough to handle with all these contacts. Is the only option here increase the private memory or there is any other kind of configuration(in redis or kamailio) that i can make to avoid this? Best Regards José Seabra -- Cumprimentos José Seabra

2 1

Re: [SR-Users] Kamailio IMS installation from pacakges
by Mohit Chawla 24 May '19

24 May '19

Hi, My name is Mohit Chawla working on LTE Network integration with Kamailio IMS server. We are facing issues in installation & Configuring of kamailio IMS on CentOS 7 due to no documentation is available on web (or at least I am not able to find one). Primarily I am interested in installing packages (not building from source code as of now) and use it for further testing. I can go with building the package from source if its more easier and quick. Can you guys help me share any document/wiki which is helpful for my cause ? I will be really grateful if someone can reply me by today. Thanks and Regards, Mohit From: Mohit Chawla [mailto:mohit.chawla@azcom.in] Sent: Wednesday, May 22, 2019 5:57 PM To: 'sr-users(a)lists.kamailio.org' Subject: Kamailio IMS installation from pacakges Hi, My name is Mohit Chawla working on LTE Network integration with Kamailio IMS server. We are facing issues in installation & Configuring of kamailio IMS on CentOS 7 due to no documentation is available on web (or at least I am not able to find one). Primarily I am interested in installing packages (not building from source code as of now) and use it for further testing. I can go with building the package from source if its more easier and quick. Can you guys help me share any document/wiki which is helpful for my cause ? I will be really grateful if someone can reply me by today. Thanks and Regards, Mohit -- _Disclaimer: This email and any files transmitted along with it may contain Azcom confidential and proprietary information. If you are not the intended recipient, you are notified that disclosing, copying, distributing or taking any action based on the contents of the information contained herein is strictly prohibited. If you are not an intended recipient of this transmission and you received it in error, please inform the sender by reply e-mail and destroy this and all other copies of this transmission to which you have access. The sender of this email or Azcom does not accept liability for any errors or omissions in the contents of this message that may occur as a result._

2 3

Use of kamailio as IMS server to integrate with LTE Core Network
by Mohit Chawla 24 May '19

24 May '19

Hi , We have successfully installed the kamailio IMS server on centOS 7 using the following link (http://www.kamailio.org/wiki/tutorials/ims/installation-howto) with appropriate changes for command in CentOS 7. We are currently trying to integrate the PCSCF module with LTE Core network , but I don't have any idea how to do this. Neither I am able to find any tutorials which I can use to understand how kamailio works. Can anyone share any reference document to help me understand the kamailio working in my use case. Thanks and Regards, Mohit -- _Disclaimer: This email and any files transmitted along with it may contain Azcom confidential and proprietary information. If you are not the intended recipient, you are notified that disclosing, copying, distributing or taking any action based on the contents of the information contained herein is strictly prohibited. If you are not an intended recipient of this transmission and you received it in error, please inform the sender by reply e-mail and destroy this and all other copies of this transmission to which you have access. The sender of this email or Azcom does not accept liability for any errors or omissions in the contents of this message that may occur as a result._

1 0

listen (bind) to variable ip and port
by Roiss-Weiss, Roman 23 May '19

23 May '19

Hi guys, is it somehow possible to bind to a variable ip-address and port? e.g. getting the ip and port from a database? Kind regards, Roman

3 2

Kamailio CDRs for calls interrupted without BYE
by Ildar Mukhametzhanov 23 May '19

23 May '19

Hello! I am using dialog+acc modules to write CDRs in kamailio. Everything works great except one test. During active call I switched off both UACs preventing them to send BYE. Rtpengine reported about call interruption, but kamailio thinks that this call is in-progress for a long time. What is the right way to make kamailio know about calls interrupted without BYE. Thanks. -- Sincerely, *Ildar Mukhametzhanov* Head of software development VoiceON Telecom Services GmbH im(a)voice-on.net <mailto:im@voice-on.net> t. +49 30 398 20 500-77

3 8

warning logs - DB_Redis and usrloc in mode 3
by José Seabra 22 May '19

22 May '19

Hello there, I'm testing the db_redis with usrloc module in mode 3 and i have been noticing that every time that kamailio checks if there is contacts expired then it logs the following warning messages: *WARNING: PR_LOG: 5a14aea50d9d9b17-75(a)172.31.1.11 <5a14aea50d9d9b17-75(a)172.31.1.11>: db_redis [redis_dbase.c:1099]: db_redis_perform_query(): performing full table scan on table 'location' while performing query* *WARNING: PR_LOG: 5a14aea50d9d9b17-75(a)172.31.1.11 <5a14aea50d9d9b17-75(a)172.31.1.11>: db_redis [redis_dbase.c:1102]: db_redis_perform_query(): scan key 0 is 'expires'* *WARNING: PR_LOG: 5a14aea50d9d9b17-75(a)172.31.1.11 <5a14aea50d9d9b17-75(a)172.31.1.11>: db_redis [redis_dbase.c:1102]: db_redis_perform_query(): scan key 1 is 'expires'* *WARNING: PR_LOG: 5a14aea50d9d9b17-75(a)172.31.1.11 <5a14aea50d9d9b17-75(a)172.31.1.11>: db_redis [redis_dbase.c:1274]: db_redis_perform_delete(): performing full table scan on table 'location' while performing delete* *WARNING: PR_LOG: 5a14aea50d9d9b17-75(a)172.31.1.11 <5a14aea50d9d9b17-75(a)172.31.1.11>: db_redis [redis_dbase.c:1277]: db_redis_perform_delete(): scan key 0 is 'expires'* *WARNING: PR_LOG: 5a14aea50d9d9b17-75(a)172.31.1.11 <5a14aea50d9d9b17-75(a)172.31.1.11>: db_redis [redis_dbase.c:1277]: db_redis_perform_delete(): scan key 1 is 'expires'* I'm not sure what is the rational behind of this, but having these logs being written when we have log level equal 2 in production is something unnecessary, so i'm asking if it shouldn't be better we have it as debug instead of warning? or do you forecast any performance issues using usrloc module in mode 3 with db_redis? Thank you for your support Cheers José -- Cumprimentos José Seabra

2 1

Re: [SR-Users] Modern SIP Trunking, Route on To: or INVITE Headers?
by JR Richardson 22 May '19

22 May '19

Benoit, Alex, I appreciate the feedback, confirming my assumptions. Good news is the Vendor finally responded positively and agreed to add in configuration option to route on INVITE R-URI in next SDK release, expected later this year. In the meantime I added 'uac_replace_to' function to my proxy to re-write To: header with R-URI, so far so good. Thanks. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope > > The To header is a cosmetic, purely logical commentary on the intended ultimate destination of the call, and no routing should ever be done on it. The RURI of the INVITE is the appropriate vehicle for that. > > When dealing with proxies, however, one will encounter from time to time the grim reality that there are user agents which expect the To URI and the INVITE RURI to be in alignment. For that scenario, you can use ‘uac’ module’s stateful rewriting functionally (uac_replace_to()) to modify To with Kamailio in a way that displays fidelity and esteem for protocol formalities. > > — > Sent from mobile, with due apologies for brevity and errors. > > > On May 21, 2019, at 9:32 AM, JR Richardson <jmr.richardson(a)gmail.com> wrote: > > > > Hey Folks, > > > > I could use some feedback to see if my mind is right on this topic.I'm > > in a discussion with a software Vendor (respectfully unnamed) with > > terminal software routing on To: header info. It's my contention any > > modern SIP software should use INVITE header to retrieve DID info and > > route from there. > > > > My reasoning is due to the To: header should contain the original > > intended DID, Extension, Name, AoR or whatever else the original SIP > > transaction wanted to contact, but being in a multi-carrier, > > multi-hop, call forwarding environment, the INVITE header contains the > > hop-to-hop true intention of where the call should route at any given > > transaction. So in the case of terminal software the To: header could > > likely be irrelevant where as the INVITE header is accurate. > > > > My most recent response from the Vendor suggest I put a SIP Proxy in > > front of the terminal software and transform the To: header to > > whatever I need for this application. Although I can do this, I'm > > concerned with breaking RFC for CANCEL and BYE transaction sent back > > from the terminal software that may not be recognized by upstream > > origination carriers. > > > > Plus modifying To: header is widely frowned upon by most folks. > > > > Thanks. > > > > JR

1 0

Kamailio v5.2.3 Released
by Daniel-Constantin Mierla 22 May '19

22 May '19

Hello, Kamailio SIP Server v5.2.3 stable release is out. This is a maintenance release of the latest stable branch, 5.2, that includes fixes since the release of v5.2.2. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.2.x. Deployments running previous v5.2.x versions are strongly recommended to be upgraded to v5.2.3. For more details about version 5.2.3 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2019/05/kamailio-v5-2-3-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Kamailio IMS installation from pacakges
by Mohit Chawla 22 May '19

22 May '19

Hi, My name is Mohit Chawla working on LTE Network integration with Kamailio IMS server. We are facing issues in installation & Configuring of kamailio IMS on CentOS 7 due to no documentation is available on web (or at least I am not able to find one). Primarily I am interested in installing packages (not building from source code as of now) and use it for further testing. I can go with building the package from source if its more easier and quick. Can you guys help me share any document/wiki which is helpful for my cause ? I will be really grateful if someone can reply me by today. Thanks and Regards, Mohit -- _Disclaimer: This email and any files transmitted along with it may contain Azcom confidential and proprietary information. If you are not the intended recipient, you are notified that disclosing, copying, distributing or taking any action based on the contents of the information contained herein is strictly prohibited. If you are not an intended recipient of this transmission and you received it in error, please inform the sender by reply e-mail and destroy this and all other copies of this transmission to which you have access. The sender of this email or Azcom does not accept liability for any errors or omissions in the contents of this message that may occur as a result._

1 0

Releasing Kamailio v5.2.3
by Daniel-Constantin Mierla 22 May '19

22 May '19

Hello, I am considering to release Kamailo v5.2.3 out of latest git branch 5.2 next week, likely on Wednesday, May 22. As usual, if you are aware of issues not reported to the github bug tracker, fill a report there as soon as possible to give a chance to be analyzed and fixed. Also, if there was a fix for an issue that you were reporting, check to see if it was backported to be sure it is not forgotten. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 1

remove_hf inside of onreply_route
by Володимир Іванець 21 May '19

21 May '19

Hello! I was wondering if someone is using remove_hf in onreply_route and if yes on which Kamailio version? I'm using Kamailio v5.2.1 and as soon as I add remove_hf function to onreply_route Kamailio will fail to start with following error messages: ERROR: <core> [core/mem/q_malloc.c:291]: qm_find_free(): qm_find_free(0x7fc1dc044010, 27120); Free fragment not found! ERROR: <core> [core/mem/q_malloc.c:425]: qm_malloc(): qm_malloc(0x7fc1dc044010, 27120) called from core: core/io_wait.c: init_io_wait(524), module: core; Free fragment not found! Thank you for any suggestions!

2 2

Modern SIP Trunking, Route on To: or INVITE Headers?
by JR Richardson 21 May '19

21 May '19

Hey Folks, I could use some feedback to see if my mind is right on this topic.I'm in a discussion with a software Vendor (respectfully unnamed) with terminal software routing on To: header info. It's my contention any modern SIP software should use INVITE header to retrieve DID info and route from there. My reasoning is due to the To: header should contain the original intended DID, Extension, Name, AoR or whatever else the original SIP transaction wanted to contact, but being in a multi-carrier, multi-hop, call forwarding environment, the INVITE header contains the hop-to-hop true intention of where the call should route at any given transaction. So in the case of terminal software the To: header could likely be irrelevant where as the INVITE header is accurate. My most recent response from the Vendor suggest I put a SIP Proxy in front of the terminal software and transform the To: header to whatever I need for this application. Although I can do this, I'm concerned with breaking RFC for CANCEL and BYE transaction sent back from the terminal software that may not be recognized by upstream origination carriers. Plus modifying To: header is widely frowned upon by most folks. Thanks. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope

4 3

$snd(proto) mapping?
by Andrew Chen 21 May '19

21 May '19

Hey guys, Is there a numeric mapping for $snd(proto)? Thanks -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

2 1

How to get started (configuring an interconnection)
by Benoit Panizzon 21 May '19

21 May '19

Dear Kamailio Users I might need a bit help with starting. We are a TSP and at the moment we use a commercial carrier grade voice switch, which is very VERY inflexible and where bugs we find either don't get fixed, or only with a large delay. As everywhere else we embrace opensource and enjoy the possibility to quickly fix bugs ourselves (we have a software development department) we are evaluating Kamailio. Basically, if we have a centralized SIP routing engine, which can record CDRS for Billing and do some Number Translations (lookup on external service or database) for (location based emergency numbers, call forwarding services, ported numbers, call spam blacklisting) we could do way more than our actual $$$ carrier switch is capable of. Kamailio sounds like perfect for this task. First step: Get a machine up and running with Kamailio and Siremis, done, that was easy. Next: Connect his SIP wise via what I would call a 'SIP carrier interconnecton', ip based authenticated trunk between our carrier switch and Kamailio so we can route some number ranges from our carrier switch to Kamailio and then continue to test with subscribers, subscriber trunks and all else we need on Kamailio. I'm quite fluent in Asterisk (our Voicemail and Announcement Services are based on Asterisk) and SIP and have come across many different PBX, but after reading parts of the documentation, and trying myself to understand how Kamailio talks to other SIP endpoints I am a bit at a loss. I would have expected that I would configure our carrier switch as 'remote SIP gateway' or whatever it would be called. But I start fearing, Kamilio works completely differently. So I might need a bit of pointing the right direction to get started. I did try with the 'CarrierRoute Management' or 'Dispatcher' functions, but I only find very little documentation on this. So how do I get started? How do I, for starters, tell Kamailio what the Hostname or IP of my Carrier switch is and how do I set a 'default' call route to this? Kind regards -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 1

Storing an integer value into AVP from REDIS string
by Stephen Bucklin 21 May '19

21 May '19

Hi, Any quick thoughts. I am using a REDIS "GET" to retrieve a JSON obj as a string from REDIS and then the JANSSON get value to extract a key value from the string and pop it into an AVP ($avp(val)) and wish to ensure the value is an integer in the AVP. New to this, be kind :-) Steve

4 6

How to define IP Trunk in kamailio
by Mary Honey 21 May '19

21 May '19

Hello everyone, I install and setup an IMS Network by kamailio-5.2.2 . I have p-cscf, i-cscf and s-cscf in three different systems . my Hss is Fokus. Registration and call between subscribers of network work successfully . now I want to define IP Trunk in my network which can pass external calls through my IMS network. I don't know how to define IP Trunk and changes in which element(pcscf, icscf or scscf) should be done.

2 3

Rewrite contact header
by R, Rajkumar (Raj) 21 May '19

21 May '19

Hello, We are using alias routing from NATHelper module and I've a case where the alias part 'alias=ip~port~transport' needs to be removed from the contact header in 2xx message. There's contact header reference "$ct" pseudo variable but it is read only. Is there any approach available to achieve this as I haven't found anything in the forum archive search so far. thanks, raj

2 2

Unable to CDRs on "Busy Here 486" from remote side and save start and end time on successful call in table
by Shahid Mehmood 21 May '19

21 May '19

Hello, i want to save the status each call into the database. but failed to 1. record CDRs for "Busy Here 486/ service unavailable 503 " from remote side in database. 2. on successful call unable to record total time of call in database. Version of Kamailio Is *version: kamailio 4.4.7 (x86_64/linux) 35d8abflags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RESADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MBpoll method support: poll, epoll_lt, epoll_et, sigio_rt, select.id: 35d8ab* *Also Attached Config File* *Thanks*

1 0

kamailio Early IMS security
by eyas barhouk 20 May '19

20 May '19

hello community Does kamailio support Early IMS security authentication schema to be able to integrate with Huawei HSS ?? and if it's does how to configure it? B.R Eyas

1 0

Plans for next major release - v5.3.0
by Daniel-Constantin Mierla 20 May '19

20 May '19

Hello, a while ago, during the last IRC devel meeting, it was proposed to either do the next major release (5.3.0) before the usual summer holidays or leave it for autumn. Given that time is approaching, we have to decide. If it is going to happen before the summer holidays, probably we should freeze the development in 2-3 weeks, then do 3-4 weeks of testing and release during the first part of July. It would be a shorter development cycle, so I do not expect much needs for testing. Only a few new modules and plenty of work to consolidate and enhance the existing code. Any preference for one of the two options? Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

3 2

Kamailio usrloc mode 3 using mongodb
by José Seabra 20 May '19

20 May '19

Hello there, I'm testing usrloc module with mongodb and i have a couple of questions to do. What i would like to know/have confirmation is if the fields that are specified in the following link https://kamailio.org/docs/db-tables/kamailio-db-5.2.x.html#gen-db-location-… can be used to create indexes in mongodb? I'm noticing that when a register is updated/deleted or expires kamailio prints the following INFO logs: *db_mongodb [mongodb_dbase.c:526]: db_mongodb_get_columns(): unhandled data type column (methods) type id (10), use DB1_STRING as default* *db_mongodb [mongodb_dbase.c:526]: db_mongodb_get_columns(): unhandled data type column (instance) type id (10), use DB1_STRING as default* These fields (*methods and **instance*) are stored in mongodb document with a null string, e.g: *{* * "_id" : ObjectId("5cdc775a6472740055289a11"),* * "username" : "x1000",* * "contact" : "sip:x1000@192.168.1.1:5674;transport=UDP",* * "expires" : ISODate("2019-05-15T20:33:26Z"),* * "q" : -1,* * "callid" : "sipp-ci2-edge-sip-proxy1-20190515203220-1-21(a)172.31.2.61 <sipp-ci2-edge-sip-proxy1-20190515203220-1-21(a)172.31.2.61>",* * "cseq" : 2,* * "flags" : 0,* * "cflags" : 0,* * "user_agent" : "sipp",* * "received" : "sip:191.30.10.23:8888 <http://191.30.10.23:8888>",* * "path" : "<sip:172.31.2.61:5060;received=sip:191.30.10.23:8888;lr>",* * "socket" : "udp:172.31.2.11:5060 <http://172.31.2.11:5060>",* * "methods" : null,* * "last_modified" : ISODate("2019-05-15T20:32:26Z"),* * "ruid" : "uloc-15-5cdc7756-55-1",* * "instance" : null,* * "reg_id" : 0,* * "server_id" : 21,* * "connection_id" : -1,* * "keepalive" : 0,* * "partition" : 0* *}* Should i be concerned about that log msg? how can i avoid it? Thanks Regards -- Cumprimentos José Seabra

2 5

BYE relays Digest and others in Proxy-Authorization header
by Sergiu Pojoga 19 May '19

19 May '19

Hi ppl, Noticed BYE requests relayed from caller to callee have the *Proxy-Authorization* header including *Digest*, *nonce* and *response*. Is that by SIP definition or am I doing something wrong? My concern is the callee gets to see caller's authentication credentials. Thanks.

4 6

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